Revert 240548 "Enable platform echo cancellation through the Aud..."

trunk/src/content/renderer/media/webrtc_audio_capturer.cc

Index: trunk/src/content/renderer/media/webrtc_audio_capturer.cc
===================================================================
--- trunk/src/content/renderer/media/webrtc_audio_capturer.cc	(revision 240588)
+++ trunk/src/content/renderer/media/webrtc_audio_capturer.cc	(working copy)
@@ -114,8 +114,7 @@
 }
 
 void WebRtcAudioCapturer::Reconfigure(int sample_rate,
-    media::ChannelLayout channel_layout,
-    int effects) {
+                                      media::ChannelLayout channel_layout) {
   DCHECK(thread_checker_.CalledOnValidThread());
   int buffer_size = GetBufferSize(sample_rate);
   DVLOG(1) << "Using WebRTC input buffer size: " << buffer_size;
@@ -125,8 +124,9 @@
 
   // bits_per_sample is always 16 for now.
   int bits_per_sample = 16;
-  media::AudioParameters params(format, channel_layout, 0, sample_rate,
-                                bits_per_sample, buffer_size, effects);
+  media::AudioParameters params(format, channel_layout, sample_rate,
+                                bits_per_sample, buffer_size);
+
   {
     base::AutoLock auto_lock(lock_);
     params_ = params;
@@ -137,14 +137,13 @@
 }
 
 bool WebRtcAudioCapturer::Initialize(int render_view_id,
-    media::ChannelLayout channel_layout,
-    int sample_rate,
-    int buffer_size,
-    int session_id,
-    const std::string& device_id,
-    int paired_output_sample_rate,
-    int paired_output_frames_per_buffer,
-    int effects) {
+                                     media::ChannelLayout channel_layout,
+                                     int sample_rate,
+                                     int buffer_size,
+                                     int session_id,
+                                     const std::string& device_id,
+                                     int paired_output_sample_rate,
+                                     int paired_output_frames_per_buffer) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
 
@@ -210,8 +209,7 @@
   // providing an alternative media::AudioCapturerSource.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     channel_layout,
-                    static_cast<float>(sample_rate),
-                    effects);
+                    static_cast<float>(sample_rate));
 
   return true;
 }
@@ -284,8 +282,7 @@
 void WebRtcAudioCapturer::SetCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
-    float sample_rate,
-    int effects) {
+    float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
            << "sample_rate=" << sample_rate << ")";
@@ -311,7 +308,7 @@
   // Dispatch the new parameters both to the sink(s) and to the new source.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
-  Reconfigure(sample_rate, channel_layout, effects);
+  Reconfigure(sample_rate, channel_layout);
 
   // Make sure to grab the new parameters in case they were reconfigured.
   media::AudioParameters params = audio_parameters();
@@ -350,8 +347,7 @@
   // WebRtc native buffer size.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
                     params.channel_layout(),
-                    static_cast<float>(params.sample_rate()),
-                    params.effects());
+                    static_cast<float>(params.sample_rate()));
 }
 
 void WebRtcAudioCapturer::Start() {